<pre>The phone is only making one call, notice the call-id did not change. The second INVITE is sent in responce to a 401 Authentication Required. The 401 will contain the necessary authentication information for the phone to use to build the Authorization header that it inserts in the second invite. THe mechanism uses a shared secret (the reg.X.auth.userId and reg.X.auth.password in the polycom cfg file, and the secret="XXXXX" and the userID(I think thats what its called) in the asterisk config files). <br>
<br>If you have other phones that are not doing this second invite I would bet its because on the asterisk side you have not configured them to use a secret.<br><br>----------------------------------------------------------------------------------<br>
Thanks for the tip, I did just that, and now I am more confused.<br><br>It does appear as though there is just one call ID (if my assumption that the "tag=" determines the call.<br><br>The first time it sends like this:<br>
<br><--- SIP read from UDP:x.x.x.x:5060 ---><br>INVITE <a class="moz-txt-link-freetext" href="sip:3261@y.y.y.y;user=phone">sip:3261@y.y.y.y;user=phone</a> SIP/2.0<br>Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bKe3e15c76913F8BDD<br>
From: "3271" <a class="moz-txt-link-rfc2396E" href="sip:3271@y.y.y.y"><sip:3271@ y.y.y.y ></a>;tag=990EE6B0-8E3DCEA7<br>To: <a class="moz-txt-link-rfc2396E" href="sip:3261@y.y.y.y;user=phone"><sip:3261@ y.y.y.y;user=phone></a><br>
CSeq: 1 INVITE<br>Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:96a1fe9c-88f06c73-7e209322@x.x.x.x">96a1fe9c-88f06c73-7e209322@x.x.x.x</a><br>Contact: <a class="moz-txt-link-rfc2396E" href="sip:3271@x.x.x.x:5060"><sip:3271@ x.x.x.x:5060></a><br>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER<br>User-Agent: PolycomSoundStationIP-SSIP_6000-UA/3.2.3.1734<br>Accept-Language: en<br>Supported: 100rel,replaces<br>Allow-Events: talk,hold,conference<br>
Max-Forwards: 70<br>Content-Type: application/sdp<br>Content-Length: 461<br><br>v=0<br>o=- 1271881915 1271881915 IN IP4 x.x.x.x<br>s=Polycom IP Phone<br>c=IN IP4 x.x.x.x<br>t=0 0<br>a=sendrecv<br>m=audio 2226 RTP/AVP 115 99 9 102 0 8 18 127<br>
a=rtpmap:115 G7221/32000<br>a=fmtp:115 bitrate=48000<br>a=rtpmap:99 SIREN14/16000<br>a=fmtp:99 bitrate=48000<br>a=rtpmap:9 G722/8000<br>a=rtpmap:102 G7221/16000<br>a=fmtp:102 bitrate=32000<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>
a=rtpmap:18 G729/8000<br>a=fmtp:18 annexb=no<br>a=rtpmap:127 telephone-event/8000<br><br>Asterisk responds with a SIP/2.0 401 Unauthorized, the phone then comes back with this:<br><br><--- SIP read from UDP:x.x.x.x:5060 ---><br>
INVITE <a class="moz-txt-link-freetext" href="sip:3261@">sip:3261@</a> y.y.y.y;user=phone SIP/2.0<br>Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK6f7a6692AF94008<br>From: "3271" <a class="moz-txt-link-rfc2396E" href="sip:3271@y.y.y.y"><sip:3271@ y.y.y.y ></a>;tag=990EE6B0-8E3DCEA7<br>
To: <a class="moz-txt-link-rfc2396E" href="sip:3261@y.y.y.y;user=phone"><sip:3261@ y.y.y.y;user=phone></a><br>CSeq: 2 INVITE<br>Call-ID: 96a1fe9c-88f06c73-7e209322@ x.x.x.x<br>Contact: <a class="moz-txt-link-rfc2396E" href="sip:3271@x.x.x.x:5060"><sip:3271@ x.x.x.x:5060></a><br>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER<br>User-Agent: PolycomSoundStationIP-SSIP_6000-UA/3.2.3.1734<br>Accept-Language: en<br>Supported: 100rel,replaces<br>Allow-Events: talk,hold,conference<br>
Authorization: Digest username="3271", realm="asterisk", nonce="393a1b1f", uri=<a class="moz-txt-link-rfc2396E" href="sip:3261@y.y.y.y;user=phone">"sip:3261@ y.y.y.y;user=phone"</a>, response="c8223e261c252c12172982ee661ad307", algorithm=MD5<br>
Max-Forwards: 70<br>Content-Type: application/sdp<br>Content-Length: 461<br><br>v=0<br>o=- 1271881915 1271881915 IN IP4 x.x.x.x<br>s=Polycom IP Phone<br>c=IN IP4 x.x.x.x<br>t=0 0<br>a=sendrecv<br>m=audio 2226 RTP/AVP 115 99 9 102 0 8 18 127<br>
a=rtpmap:115 G7221/32000<br>a=fmtp:115 bitrate=48000<br>a=rtpmap:99 SIREN14/16000<br>a=fmtp:99 bitrate=48000<br>a=rtpmap:9 G722/8000<br>a=rtpmap:102 G7221/16000<br>a=fmtp:102 bitrate=32000<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>
a=rtpmap:18 G729/8000<br>a=fmtp:18 annexb=no<br>a=rtpmap:127 telephone-event/8000<br><br><br>The difference is that the CSeq is now 2 and the following line is added:<br><br>Authorization: Digest username="3271", realm="asterisk", nonce="393a1b1f", uri=<a class="moz-txt-link-rfc2396E" href="sip:3261@y.y.y.y;user=phone">"sip:3261@y.y.y.y;user=phone"</a>, response="c8223e261c252c12172982ee661ad307", algorithm=MD5<br>
<br><br>So maybe I do have an issue in Asterisk, okay probably. Any clues as to how to debug? Let me know if need to post more information.<br><br>Thanks.<br><br>-Jay<br><br>-----Original Message-----<br>From: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> [<a class="moz-txt-link-freetext" href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Sean Brady<br>
Sent: Tuesday, April 20, 2010 4:57 AM<br>To: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>Subject: Re: [asterisk-users] Odd Issue With Polycom Phones<br>
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